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...1 make: *** [scripts_basic] Error 2 error: Bad exit status from /var/tmp/rpm-tmp.50765 (%prep) RPM build errors: Bad exit status from /var/tmp/rpm-tmp.50765 (%prep) Does any other package needs to be installed for fixing this? gcc, glibc-devel, gcc-c++ are already installed. Regards, Kurian Thayil. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20090629/211aa0dd/attachment.html>

...ne with option 't' in the queue() application and I was able to successfully transfer calling party (client) to another agent. But is it possible for the new agent to transfer this calling party to another agent? ie Does a second transfer is supported in a Queue? Regards, -- Kurian Mathew Thayil. (GPG KeyID: E232394F) -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090530/468705a9...

...error I understand this is an error message from Error Detection And Control module and just wanna confirm that this is not a kernel or software related issue. If Hardware related is it confined only to the Physical memory stick installed or processor related? Any hint on this?? Regards, Kurian Thayil. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20090704/ab749883/attachment-0003.html>

...e, SIP/8440 rings. Is there any other way to implement this. I am not ready to use Vicidial (AstGUIClient) because the interface to be designed is too custom and the agent should have the list of numbers in front of them while they dial which cannot be done using Vicicial. Regards, Kurian Mathew Thayil. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090310/947be093/attachment.htm

...ile read -e ARG && [ "$ARG" ] ; do array=(` echo $ARG | sed -e 's/://'`) echo ${array[0]} = ${array[1]} >>$LOG_FILE export ${array[0]}=${array[1]} done echo "EXEC Dial SIP/$agi_extension 20 tTo" *Any hint on this?? Regards, Kurian Thayil. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090520/5ba2b86e/attachment.htm

Hi, Which is the recommended filesystem to be used for the bricks in glusterFS. ?? XFS/EXT3/EXT4 etc .???? Thanks & Regards, Bobby Jacob Senior Technical Systems Engineer | eGroup P SAVE TREES. Please don't print this e-mail unless you really need to. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Is there a way that we can add custom IP-Tables rules in a nat'd physical host? I need some custom rules mentioned in physical host to access some services in the guest systems. Any hints on this? Regards, Kurian Thayil. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://listman.redhat.com/archives/libvirt-users/attachments/20110409/c85547ae/attachment.htm>

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]

Hello(s), I know this might be test book question or one best suited for google but I will take the risk of asking. Here I go. What common routine maintenance tasks do you run on your asterisk box? Thanks James. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090422/8cfeb434/attachment.htm

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Hi All, Is there a way that I can bring the callerID obtained from an incoming call to display in a web browser. I am using eyeBeam as softphone. Please help me with this. Thanks in advance. Regards, Kurian Thayil. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090506/7de6e24c/attachment.htm

...new stack Here, from above, AGI program agi_qdial.sh which handles the dial operation does not make any attempt to dial 85766. Wondering why this is happening. The issue gets resolved only when asterisk service is restarted which is not a pretty good workaround. Any clue on this? Regards, Kurian Thayil. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090717/167d706d/attachment.htm

Hi All, I created a couple of virtual networks (forward mode=nat) in my rhel6-kvm box. I've come across 2 weird issues. 1. My Iptables rule chainset contains repeated rules. The same rule gets repeated block by block 2. For connecting to guest using SSH, I created a custom IPTables chain. I want this chain to be on top of the FORWARD chain, but everytime the libvirtd is restarted the rule